In U.S. Pat. No. 5,940,610, there is described the use of prioritized interrupt callback routines to process different types of multimedia information. Here, multimedia information passes through a system bus from a CPU main memory to a display memory in accordance with CPU commands. The information may be packetized with associated packet types identifying the different media. A media stream controller processes the information and passes the processed information to the display memory. Controllers in the media stream controller individually pass multimedia information to the display memory. A PACDAC controller in the media stream controller causes media in the display memory to be transferred to a PACDAC for display. The format, sequence, and rate of this transfer may be flexibly controlled by software on a frame by frame basis. Arbitration logic establishes priorities for the different controllers in the media stream controller so they may share a single bus for accessing the display memory. A single interrupt controller coordinates interrupts to provide priorities based upon the type of interrupt cause or media. Each interrupt cause activates only the appropriate callback functions. Two virtual machine sessions share an interrupt line to process interrupt requests from one session before processing interrupt requests from the other.
In US 2002/0099842 A1, there is described a system and method for the routing of media. The system and method for streaming media to a viewer and managing the media comprises an enhanced service routing processor, a real time switch management system, a name routing processor, and a managed media switch. The real time switch management system RTSMS has a reservation system. The enhanced service routing processor ESRP receives media from an owner, manages the media according to media rules and order rules defined by the owner, and distributes the media to one or more switches, such as the managed media switch MMS, according to the media rules and the order rules.
With the fast improvement of broadband wireless technologies, more and more demanding multimedia services are provided over wireless access communication networks. However, the highly variable nature of wireless radio communication links and the special error characteristics require special practices not just at the physical and link layer of the networking protocol stack, but also on higher layers thereof. In other words, with the spreading of wireless access technologies, the paradigm of homogeneous networks, where all links are similar in terms of delay and error probability and substantially static in nature, does not hold any longer. The wireless communication network and especially the applications using such networks must be prepared for the special properties of heterogeneous wired-cum-wireless networks.
In heterogeneous wireless communication networks, real-time service delivery is a highly non-trivial task in view of varying transport conditions which makes it difficult to meet strict delay constraints. To handle the challenging environment, the special characteristics of real-time services must be fully exploited. With real-time service delivery, unlike TCP-based applications like file download which demand lossless data transmission, a certain level of data loss is acceptable. For real-time service delivery data is assumed to be lost when packets arrive not or only too late at end user terminals. Traditionally, communication networks assume equal loss importance for each data packet and all the data in a single data packet is assumed to be equally important, e.g., each bit of a data packet has the receiving probability.
For real-time multimedia and audiovisual data traffic, this assumption does not hold. In general, consecutive data packets of a media stream carry data of different importance for user-perceived quality. Also, contents of each data packet may be related to data of various importance to achieve a certain play-out quality. E.g., although a data loss due to congestion or bad wireless communication conditions might be tolerated, it still matters what type of data is actually lost.
Thus, to cope with the above problems, there have been suggested techniques like rate adaptation RA and unequal error protection UEP, which provide means to exploit the benefits of data characteristics for real-time service delivery.
Here, with rate adaptation RA, the decrease or increase in the source bit-rate may be controlled. Hereby, during congestion or a degradation of the wireless communication the transmitted bit-rate of an application and hence the perceived quality of audiovisual data may be reduced in order to maintain an acceptable quality for service delivery.
With unequal error protection UEP, the varying importance of data can be taken into account when applying an error protection scheme for transmission during service delivery. I.e., from the viewpoint of the application, important data parts receive a stronger error protection than the less important parts.
Third generation networks like the universal mobile telecommunication system UMTS have been developed to support high data rates to permit access to a wide selection of services, besides circuit switched CS transport 3G networks implement packets switched PS data transport. Traditionally, circuit switched transport is used for voice-telephony, while packet switched traffic is used for data traffic delivery. Because circuit switched transport is highly optimised for telephony, rate adaptation RA and unequal error protection UEP are fully exploited.
Further, by realising the benefits of a packet switched transport, real-time service delivery is shifting from circuit switched transport to packet switched transport. To extend the capability of packets switched transport allowing for sophisticated rate adaptation and unequal error protection techniques, currently a narrow set of service adaptation is possible in comparison with the circuit switched domain.
With circuit switched voice-telephony, the adaptive multi-rate AMR codec, “AMR speech codec; general description”, Technical Report 26.071, 3GPP, June 2002, is tightly coupled to the transport network through sophisticated adaptation mechanisms. E.g., during bad radio conditions, the data-rate of the adaptive multi-rate codec is reduced permitting the use of a stronger error protection. The radio access network RAN provides the means for this adaptation in case of circuit switched transmission, while no similar mechanisms exist for packet switched transport, although scalable codecs, unequal error protection UEP, and rate adaptation RA methods would be desirable also for packet switched based multimedia service delivery.
In more detail, with respect to packet switching, since for data transmission over a packet switched bearer the radio network controller RNC and the core network elements, serving GPRS support node SGSN and gateway GPRS support node GGSN are service agnostic, the payload format will not be interpreted by these network elements.
Also, in the circuit switched operation mode, the speech parameter bits delivered by the adaptive multi-rate AMR codec are rearranged according to their subjective importance before they are sent to the communication network. The rearranged speech parameter bits are further sorted, based on their sensitivity to errors and then are also divided into classes of importance.
Also, with respect to unequal error protection, the split over different blocks of priority is necessary. Hence, more sensitive speech bits are protected to a higher extent than less sensitive speech bits, to guarantee a certain perceived quality at the receiver.
Further, the length of the encoded speech payload may vary depending on the speech codec mode. Before transmission of the speech frames over the wireless communication channel, the splitting and reassembling of the adaptive multi-rate AMR encoded speech payload is done by the radio network controller RNC and the user equipment UE, respectively. Hence, the radio network controller RNC and the user equipment UE need the exact payload format and the block length information with respect to each priority class.
A solution could be that the radio network controller RNC becomes service aware also for packet switched transmission which, however, would have the drawback that the format and length of service data units SDU, generated by the source codec for each priority class, need to be downloaded to the radio network controller RNC, e.g., using the UMTS QoS profile description.
In this case, the problem is that the radio network controller will lose service transparency through handling of information generated on the application level. Moreover, by relaxing service-awareness of network elements, system architectures can be exploited in a simpler way, as fewer specialized equipments would be required. The need for upgrading network elements for launching new services is reduced to a minimum and might even become obsolete. This way, by relaxing service-awareness, cheaper network deployment and better service integration may be achieved.